<div>Thank you very much for your reply . I asked on the users mailing list and waited for a long time but didn't receive any response . So I thought if it must be asked from the developers mailing list . Sorry for my mistake .</div>
<div>Thank you in advance</div>
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<div class="gmail_quote">On Tue, Dec 15, 2009 at 8:07 PM, John Todd <span dir="ltr"><<a href="mailto:jtodd@digium.com">jtodd@digium.com</a>></span> wrote:<br>
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<div class="h5"><br>On Dec 15, 2009, at 4:02 AM, hadi motamedi wrote:<br><br>> ---------- Forwarded message ----------<br>> From: hadi motamedi <<a href="mailto:motamedi24@gmail.com">motamedi24@gmail.com</a>><br>
> Date: Sat, Dec 12, 2009 at 9:44 AM<br>> Subject: Inquiry:Asterisk sip server?<br>> To: Asterisk Users Mailing List - Non-Commercial Discussion <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
> ><br>><br>><br>> Dear All<br>> I have an application that calls for Asterisk sip configuration to<br>> be able to communicate with external sip server . My Asterisk 3.1.14<br>> has been installed on Debian 3.1 server and the external sip server<br>
> is @<a href="http://192.168.0.10/" target="_blank">192.168.0.10</a> , the same subnet as my Debian server<br>> @<a href="http://192.168.0.2/" target="_blank">192.168.0.2</a> . At now , the configuration is in such a way that the<br>
> call attempts reaching to my Asterisk are being routed internally ,<br>> based on my Asterisk extensions.conf settings . I need to change the<br>> current configuration in such a way that the voip call attempts to<br>
> be routed toward the external sip <a href="mailto:server@192.168.0.10">server@192.168.0.10</a> for the call<br>> routing purposes . Can you please help me how I am expected to<br>> modify my Asterisk configuration to do the job ?<br>
> Regards<br>> H.Motamedi<br><br><br></div></div>Hadi -<br> I would suggest asking this question on the "asterisk-users"<br>mailing list, as the asterisk-dev list is for discussion only of<br>internal programming, code, bug or other development-related issues.<br>
You will find that the asterisk-users list also has a much larger<br>number of active participants who may have suggestions on your<br>problem. Thanks!<br><br>(As a side note, you may wish to create a small web page which<br>
includes your configuration files and perhaps a drawing of what you<br>wish to accomplish, as I suspect the description you have given is not<br>sufficient for someone to fully answer your question. Include the<br>link to that web page description in your post to asterisk-users.)<br>
<br>JT<br><br>---<br>John Todd <a href="mailto:email%3Ajtodd@digium.com">email:jtodd@digium.com</a><br>Digium, Inc. | Asterisk Open Source Community Director<br>445 Jan Davis Drive NW - Huntsville AL 35806 - USA<br>
direct: +1-256-428-6083 <a href="http://www.digium.com/" target="_blank">http://www.digium.com/</a><br><br><br><br><br>_______________________________________________<br>--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--/" target="_blank">http://www.api-digital.com--</a><br>
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